Each year sound companies spend billions of dollars on audio technologies and audio research to find new ways to improve audio quality in performance settings. Very often sound systems are designed to be used in a specific environment. For example, in a vehicle or private room setting, audio manipulation and output quality techniques and technologies are either prescriptive or adaptive—neither of which require the need for audio engineering professionals. However, in other situations such as at a concert venue, a wide array of audio professionals must be employed. This can include monitor engineers, system technicians, and front-of-house engineers. These professionals operate mixing consoles and audio control units to produce desirable, high-quality audio output.
Whether prescriptive or adaptive, manned or unmanned, perceived sound quality is a function of complex transducer-based technologies and acoustic treatment that are typically controlled, managed and manipulated by humans, and/or audio software and hardware. As such, both human and physical capital are required to produce first-rate sound quality. However, even when the necessary human and capital has been spent, it can still be very difficult to effectively manage audio outputs in real-time. This is due to improper calibrations of signal propagation and signal degradation, as well as unwanted harmonics and soundwave reflections.
Particularly in an outdoor setting, single-source sound systems typically produce an intermittent mix of unintelligible sounds and echoes due to a given venue's size and openness. A popular solution for addressing the echo issue is to utilize distributed sound systems. Traditional distributed sound systems are less susceptible to sound variance than single source systems. However, even when these distributed systems are used, temperature gradients and wind can still steer sound in undesirable ways.
Another issue related to the size of a performance venue is when audio and video fall out of sync. As live musical performances become more and more elaborate by including digital art and screens on-stage, it is becoming increasingly difficult to reliably sync audio and video in large venues, due to highly reverberant surfaces and long decay times.
Also impacting audio intelligibility during a live performance is crowd noise. At a live event, it is not uncommon for crowds to generate noise approaching 105 dB. When this occurs, audio engineers must manipulate the supporting sound system output so that the performance audio remains 5-8 decibels higher than the noise generated by crowd. This action results in performance sound being broadcast above 110 dB, the range where the volume of sound begins to pose danger to human listeners. Frequently, audio system operators find it difficult to granularly control the loud perception of a given individual listener while managing loudness perception for the remainder of the audience. In a case where an audio quality trade-off decision has to be made, a common industry practice is to execute a remediation plan that favors the majority of listeners while the minority of listeners are forced to suffer through it.
In other instances, when various pieces of audio equipment are slightly, or completely out-of-phase, it can be difficult for audio system operators to correct these out-of-phase issues in a short period of time.
A myriad of audio functions are necessary to provide a dynamic range of audio playback and fidelity. To meet heightened demands and address new challenges, the devices of today will not only have to handle traditional telephony voice communication and low-fidelity voice recording, but also, these devices must be capable of incorporating new hardware and software to create new functions and applications such as sensing infrasonic, ultrasonic, blue light and millimeter wave exposure and reporting, and in some cases, autonomous manipulation of audio outputs. Further, such demands create the need to process signals using ‘low-loss’ methods by moving much of the processing function away from hardware and into software optimized to do so.
Review of related technology:
Line 6, Inc. has created a ‘smart mixing system’ for non-wearable ubiquitous computing devices that enables wireless and touchscreen control of live sound system components. This is accomplished via a wired connection between standard audio hardware and a proprietary physical interface. While this system integrates and controls live sound system components via touchscreen devices, it unfortunately relies on audio engineers to operate it, and does not incorporate a listener-centric way to autonomously solve audio issues experienced by an audience.
U.S. Pat. No. 5,668,884 pertains to an audio enhancement system and method of use with a sound system for producing primary sound from at least one main loudspeaker located at a main position. The audio enhancement system comprises at least one wireless transmitter, time delay circuitry, and plural augmented sound producing subsystems. Each sound subsystem is a portable unit arranged to be carried by a person located remote from the main loudspeaker and includes a wireless receiver and an associated transducer device, e.g., a pair of stereo headphones. The transmitter broadcasts an electrical signal which is representative of the electrical input signal provided to the main loudspeaker. The broadcast signal is received by the receiver and is demodulated and amplified to drive the transducer so that it produces augmented sound substantially in synchronism with the sound arriving from the main loudspeaker. To achieve that end the time delay circuitry delays the electrical signal which is provided to the transducer for a predetermined period of time corresponding generally to the time period it takes for the primary sound to propagate through the air from the main loudspeaker to the remote location at which the person is located.
U.S. Pat. No. 7,991,171 pertains to a method and apparatus for processing an audio signal in multiple audio frequency bands while minimizing undesirable changes in tonal qualities of the audio signal by determining an initial gain adjustment factor for each audio frequency band resulting from the application of an audio processing technique. A final gain adjustment factor for each band is selected from a corresponding set of weighted or unweighted initial gain adjustment factors. The set of initial gain adjustment factors from which the final gain adjustment factor for a specified audio frequency band is obtained is derived from other audio frequency bands that have the frequency of the specified band as a harmonic frequency. Changes in audio signal level within one audio frequency band thereby affect the signal level of harmonic frequencies to decrease relative changes in volume between a fundamental frequency and its harmonics.
U.S. Pat. No. 8,315,398 pertains to a method of adjusting a loudness of an audio signal may include receiving an electronic audio signal and using one or more processors to process at least one channel of the audio signal to determine a loudness of a portion of the audio signal. This processing may include processing the channel with a plurality of approximation filters that can approximate a plurality of auditory filters that further approximate a human hearing system. In addition, the method may include computing at least one gain based at least in part on the determined loudness to cause a loudness of the audio signal to remain substantially constant for a period of time. Moreover, the method may include applying the gain to the electronic audio signal.
U.S. Pat. No. 8,452,432 pertains to a user-friendly system for real time performance and user modification of one or more previously recorded musical compositions facilitates user involvement in the creative process of a new composition that reflects the user's personal style and musical tastes. Such a system may be implemented in a small portable electronic device such as a handheld smartphone that includes a stored library of musical material including original and alternative versions of each of several different components of a common original musical composition, and a graphic user interface that allows the user to select at different times while that original composition is being performed, which versions of which components are to be incorporated to thereby create in real time a new performance that includes elements of the original performance, preferably enhanced at various times with user selected digital sound effects including stuttering and filtering. The system may also optionally comprise a visualizer module that renders a visual animation that is responsive to at least the rhythm and amplitude of the system's audio output, not only for entertainment value but also to provide visual feedback for the user.
U.S. Pat. No. 8,594,319 pertains to methods and apparatuses for adjusting audio content when more multiple audio objects are directed toward a single audio output device. The amplitude, white noise content, and frequencies can be adjusted to enhance overall sound quality or make content of certain audio objects more intelligible. Audio objects are classified by a class category, by which they can be assigned class specific processing. Audio objects classes can also have a rank. The rank of an audio object's class is used to give priority to or apply specific processing to audio objects in the presence of other audio objects of different classes.
United States Patent Publication No.: 2007/0217623 pertains to a real-time processing apparatus capable of controlling power consumption without performing complex arithmetic processing and requiring a special memory resource. The real-time processing apparatus includes an audio encoder that performs a signal processing in real time on an audio signal, a second audio encoder that performs the signal processing with a smaller throughput in real time on the audio, an audio execution step number notification unit that measures step number showing a level of the throughput in the signal processing by operating the 1st audio encoder or second audio encoder, and an audio visual system control unit that executes control so that the first audio encoder operates when the measured step number is less than a threshold value provided beforehand and the second audio encoder operates when the step number is equal to or greater than the threshold value.
United States Patent Publication No.: 2011/0134278 pertains to an image/audio data sensing module incorporated in a case of an electronic apparatus. The image/audio data sensing module comprises: at least one image sensor, for sensing an image datum; a plurality of audio sensors, for sensing at least one audio datum; a processor, for processing the image datum and the audio datum according to a control instruction set to generate a processed image data stream and at least one processed audio data stream, and combining the processed image data stream and the processed audio data stream to generate an output data stream following a transceiver interface standard; a transceiver interface, for receiving the control instruction set and transmitting the output data stream via a multiplexing process; and a circuit board, wherein the image sensor, the audio sensors and the transceiver interface are coupled to the circuit board, and the processor is provided on the circuit board.
United States Patent Publication No.: 2013/0044131 pertains to a method for revealing changes in settings of an analog control console, the method comprising: receiving a captured image of the analog control console; creating a composite image by superimposing the captured image and a live image of the analog control console; and displaying the composite image.
United States Patent Publication No.: 2013/0294618 pertains to a method and devices of sound volume management and control in the attended areas. According to the proposed method and system variants the sound reproducing system comprises: sounding mode appointment device, central station for audio signal transmittance; one or more peripheral stations for audio signal reception and playback; appliance for listener's location recognition; computing device for performing calculation concerning sounding parameters at the points of each listener's location and for performing calculation of controlling parameters for system tuning. The system can be operated wirelessly and can compose a local network.
Various devices are known in the art. However, their structure and means of operation are substantially different from the present invention. Such devices fail to provide a device that can help develop a participant's knowledge in a multitude of different subject areas, while simultaneously engaging the participant physically. At least one embodiment of this invention is presented in the drawings below and will be described in more detail herein.